## How to reproduce?
1. Refer this commit, which contains the web demo to capture screen as
video stream through RTC.
2. Copy the `trunk/research/players/whip.html` and
`trunk/research/players/js/srs.sdk.js` to replace the `develop` branch
source code.
3. `./configure && make`
4. `./objs/srs -c conf/rtc2rtmp.conf`
5. open `http://localhost:8080/players/whip.html?schema=http`
6. check `Screen` radio option.
7. click `publish`, then check the screen to share.
8. play the rtmp live stream: `rtmp://localhost/live/livestream`
9. check the video stuttering.
## Cause
When capture screen by the chrome web browser, which send RTP packet
with empty payload frequently, then all the cached RTP packets are
dropped before next key frame arrive in this case.
The OBS screen stream and camera stream do not have such problem.
## Add screen stream to WHIP demo
><img width="581" alt="Screenshot 2024-08-28 at 2 49 46 PM"
src="https://github.com/user-attachments/assets/9557dbd2-c799-4dfd-b336-5bbf2e4f8fb8">
---------
Co-authored-by: winlin <winlinvip@gmail.com>
To manage an object:
```cpp
// Before
MyClass* ptr = new MyClass();
SrsAutoFree(MyClass, ptr);
ptr->do_something();
// Now
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
```
To manage an array of objects:
```cpp
// Before
char* ptr = new char[10];
SrsAutoFreeA(char, ptr);
ptr[0] = 0xf;
// Now
SrsUniquePtr<char[]> ptr(new char[10]);
ptr[0] = 0xf;
```
In fact, SrsUniquePtr is a limited subset of SrsAutoFree, mainly
managing pointers and arrays. SrsUniquePtr is better than SrsAutoFree
because it has the same API to standard unique ptr.
```cpp
SrsUniquePtr<MyClass> ptr(new MyClass());
ptr->do_something();
MyClass* p = ptr.get();
```
SrsAutoFree actually uses a pointer to a pointer, so it can be set to
NULL, allowing the pointer's value to be changed later (this usage is
different from SrsUniquePtr).
```cpp
// OK to free ptr correctly.
MyClass* ptr;
SrsAutoFree(MyClass, ptr);
ptr = new MyClass();
// Crash because ptr is an invalid pointer.
MyClass* ptr;
SrsUniquePtr<MyClass> ptr(ptr);
ptr = new MyClass();
```
Additionally, SrsAutoFreeH can use specific release functions, which
SrsUniquePtr does not support.
---------
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. Add live benchmark support in srs-bench, which only connects and
disconnects without any media transport, to test source creation and
disposal and verify source memory leaks.
2. SmartPtr: Support cleanup of HTTP-FLV stream. Unregister the HTTP-FLV
handler for the pattern and clean up the objects and resources.
3. Support benchmarking RTMP/SRT with srs-bench by integrating the gosrt
and oryx RTMP libraries.
4. Refine SRT and RTC sources by using a timer to clean up the sources,
following the same strategy as the Live source.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: Jacob Su <suzp1984@gmail.com>
Detail change log:
1. [Simple,Refactor] Remove member fields of http entry, etc.
e34b3d3aa4
2. [Ignore] Rename source to live_source.
846f95ec96
3. [Ignore] Use directly ptr in consumer.
d38af021ad
4. [Complex, Important] Use shared ptr for live source.
88f922413a
The object relationship:
---
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. When converting RTC to RTMP, it is necessary to synchronize the audio
and video timestamps. When the synchronization status changes, whether
it is unsynchronized or synchronized, print logs to facilitate
troubleshooting of such issues.
2. Chrome uses the STAP-A packet, which means a single RTP packet
contains SPS/PPS information. OBS WHIP, on the other hand, sends SPS and
PPS in separate RTP packets. Therefore, SPS and PPS are in two
independent RTP packets, and SRS needs to cache these two packets.
---------
Co-authored-by: john <hondaxiao@tencent.com>
In an SDK that supports RTC Opus stereo, the parameter "stereo=1" may
appear. SRS (Spatial Reference System) needs to handle this correctly
and return an answer to enable WebRTC stereo support.
---------
`TRANS_BY_GPT4`
When accessing the SRS Stack, you should log in and use a token for each
request, or utilize the HTTP API with a secret Bearer token included in
every request. The SRS Stack HTTP API proxies both /api/v1 and /rtc/v1
to the SRS HTTP API while ensuring secure authentication. Additionally,
there is a console in the SRS Stack that requires the same token to
request the SRS Stack HTTP API, which is then proxied to the SRS HTTP
API.
The SRS Stack runs SRS with the HTTP API listening at 127.0.0.1:1985 on
the local loopback interface, allowing only the SRS Stack to access it
without authentication. All other users must login and access the SRS
Stack through its interface, rather than directly accessing the SRS HTTP
API within the SRS Stack.
---------
Co-authored-by: panda <542638787@qq.com>
1. Ignore SDP GROUP LS.
2. Support ice in global session info.
3. Support audio codec "OPUS" or "opus".
---------
Co-authored-by: Johnny <hellojinqiang@gmail.com>
* Remove extern SrsPps* duplicate declarations
* fix(rtmp2rtc): fix video payload type for rtmp to rtc bridge (#3041)
* Revert changes not belongs to this PR.
* Fix naming issue, follow SRS style.
* Use srs_assert instead of assert.
* Fix firefox no audio issue.
Co-authored-by: winlin <winlin@vip.126.com>
1. Cache IDR frame's rtp timestamp instead of avsync timestamp.
2. Cache clock rate calculate by sender report.
3. Using srs_rtp_seq_distance instead of direct minus.
4. Add utest of av timestamp sync when duplicated sender report.
